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    High Speed Digital Signal Processing

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    High-Speed Digital Signal Processing for Satellite Communications H. Vincent Poor and Sergio Verdi Department of Electrical Engineering Princeton University Princeton, NJ 08540 USA Abstract New digital signal processing (DSP) technology opportunities, wherein signals of large bandwidth can be digitally processed and complex digital signal processing algorithms ean be implemented, make the use of new techniques that have heretofore been limited to aux dio/voice applications now feasible for space applications. Although many efforts are being devoted to the technology aspects of this work, ongoing work by the authors is concerned with the unified consideration of the theoretical and practical aspects of high-speed digital signal processing and thei applications in space communications. In this context, this paper comprises a review of some besic high-speed digital signal processing techniques that are sp- plicable in the satellite communications domain. Specifically, this paper treats the subjects of numerically stable algorithms for high-speed DSP, adaptive signal processing techniques {or demodulation of multi-access comrmunications, interference suppression in wideband com- munications, on-board signal processing for demultiplexing and transrnultiploxing, and oves- sampling techniques applied to sigma-delta modulation and related schemes. 1 Introduction and Overview ‘The field of digital technology is dramatically evolving due to the development of higher inte- gration capacities, faster analog-to-digital (A/D) and digital-to-analog (D/A) converters, and other high-speed digital technologies. The potential applications of this new technology have a direct impact on satellite communications. First, they allow one to consider digital solutions for functional blocks that wore traditionally analog in existing satellite payload. For instance, quite promising in this context are digital solutions for a significant part of the radio-frequency (RF) front end and their associated functionel blocks (routing, downconversion, and filtering). Sec- ondly, new digital processing possiblities arise for future satellite payloads, such as beamforming for example. And finally, complex digital signal processing (DSP) algorithms for small ground stations (VSAT type or portable terminals) can now be implemented, thereby significantly im- proving ground-terminal performance. A very promising possibility is this context is the use of advanced digital signel processing for mobile comrmunications terminals (e.g. multipath cancc\- lation, fading countermeasures, multiple-acoess interference suppression, ete.). ‘These new technology opportunities, wherein signals of large bandwidth can be digitally processed and complex digital signal processing algorithms can be implemented make the use of new techniques that have heretofore been limited to audio/ voice applications now feasible for space applications. Although many efforts are being devoted to the tachnology aspects of this work, the present work is concerned with the unified consideration of the theoretical and practical aspects of high-speed digital signal processing end their applications in space communications. 7A The purpose of this paper is to provide an overview of key fundamental issues that arise in the peed DSP technology for the processing of wideband communication signals. ‘We begin, in Section 2, by focusing on the fundamental issue of general signal processing or ganization for the development of numerically stable algorithms for high-speed communications applications. In most such applications, it is likely (or even necessary) that implementation will sake place at the wide bandwidths of the front end. Since this front-end (j.e., RF) bandwidth 1s mueh larger than the information bandwidths underlying the modulated signals, some signal Processing difficulties may arise in the implementation of the required processing functions. In particular, it is widely recognized that many conventional discrete-time signal processing for- mulations become ill-conditioned when applied to continuous-time processes sampled at very fast rates relative to the underlying continuous bandwidth. Recent research has shown that the above-noted difficulties can been overcome by transforming the conventional sampled-data problems, which use a shift-operator representation of data dynamics, into problems based on divided-differenced representation of data dynamics. Signal processing formulations based on this divided-difference calculus are better conditioned numerically and their solutions are nu- merically stable for in the high-speed processing regime. Section 2 will provide a review of this methodology, which is a cornerstone of the general techniques of interest in this paper. In Section 3 and 4 we will provide broad-brush overviews of two specific examples of signal processing challenges that arise in wideband communication systems in general. The framework of this discussion will be paradigmatic, as opposed to being focused on providing a compre- hensive list of processing sub-system functionalities. Specifically, we will address briefly two general signal processing problems: demultiplexing/demodulation of wideband non-orthogonal signaling formats (such as code-division multiple-access (CDMA)), and narrowband Interference (NBI) suppression. Bach of these problems serves as a paradigm for a general class of problems thet Tequire high-speed DSP functions. The first problem is an example of a general problem thet arises in wideband demultiplexing problems - namely crosstalk, which can arise due to in- tentional non-orthogonality (as in CDMA) or due to unintentional nonideal effects arising in the implementation of wideband filter banks. The second problem serves as an example of general nartowband adaptive technology, which includes beamforming, among other problems. ‘Thus, in Section 3, we consider signal processing techniques for demultiplexing/ demodu- lation of multiuser communications. This problem is a central one in the processing of uplink satellite channels and cellular radio signals. The focus of Section 3 is on adaptive techniques for multi-user demultiplexing. Such techniques permit significant performance gains (and the at- tondant increases in system capacity) over the conventional methods currently in use. However, ‘these methods are signal-processing intensive, and thus they are a natural framework within which high-speed DSP methods can applied to produce considerable improvements in system performance, For exemple, the work presented in Section 3 serves as the basis for studying the applicability of high-speed signal processing methodologies for the cancellation of the multiple access interference and the mitigation of the mobile channel and the multipath degradation effects. The adaptive multiuser framework described in Section 3 is thus a natural one for addressing these issues. Similarly, in Section 4, we give a brief review of the problem of NBI mitigation in direct- seqnience spread-spectrum (DSSS) signaling systems. A chief advantage of using a wideband for- mat such as DSSS in radio networks is that such signals ean share bandwidth with narrowband communication signals. In particular, the low spectral density of spread-spectrum signals does not interfere unduly with conventional narrowband communications; and, conversely, spread spectrum communications is inherently resistant to the NBI caused by co-existence with con ventional commmunications. However, it has been demonstrated that the performance of spread- spectrum systems in the presence of narrowband signals can be enhanced significantly through 7.2 i | i i the use of active NBI suppression prior to despreading, and Section 4 reviews this methodol- ogy. As noted above, this area serves as a good example of the general area of narrowband adaptivity, which is of general interest in on-board protessing functions such as beamforming and narrowband filtering. These functions are natural ores to implement with high-speed DSP technology. Section 5 provides a very brie? overview of some recent advances in the application of DSP teclmiques for bulk on-board wideband demultiplexing and transmultiplexing. ‘These problems respresent. more specific examples of the kind of subsystems to which the methods of Section 2 are applicable. In particular, for such a digital demultiplexers there is an inherent oversampling factor in many parts of the circuitry if @ per block approach ie implemented due to the diserep- ancy between the total bandwidth to be processed and the underlying single-user bandwidth. Thus, the framework of Section 2 can be exploited to devise effective processing architectures inal section, Section 6, we address the general issue of oversampling techniques. Such techniques rely on the tradeoff between temporal resolution (i.e., sampling speed) and amplitude resolution (i.e., quantization), to achieve greater A/D resolution than that imposed by the matching tolerances of VESI technologies. Since very high and accurate clock frequencies are straightforward to generate with current technology, this tradeoff is one that can be practically effected. The oversampling that is an a fortiori aspect of such methods makes the fundamental issues of Section 2 of particular interest in this framework. In Section 6, this methodology is teviewed in the contexts of 2 — A modulation and data modulation. 2 Numerically Stable Algorithms for High-speed DSP In the application of DSP in wideband space communications, it is likely (or even necessary) that implementation will take place at the wide bandwidths of the front end. Since this front-end (ie., RF) bandwidth is much larger than the information bandwidths underlying the modulated and interfering signals, some signal processing difficulties may arise in the implementation of the required processing functions. In particular, it is widely recognized that many conventional discrete-time signal processing formulations become ill-conditioned when applied to continuous- time processes sampled at very fast rates relative to the underlying continuous bandwidth. For example, two algorithms in which this phenomenon arises are two of the most widely used signal processing techniques - namely, Kalman-Bucy filtering and autoregressive modeling/prediction {72]. Since these and other algorithms arise frequently in communications signal processing, this issue is of considerable concern in the design of algorithms for DSP in high-bandwidth systems. In this section, we will focus on the fundamental issue of general signal processing archi- tecture for the development of mumerically stable algorithms for high-speed communications applications. In particular, recent research has shown that the above-noted difficulties can been overcome by transforming the conventional sampled-data problems, which use 2 shift-operalor representation of data dynamics, into problems based on divided-differenced representation of data dynamics. Signal processing formulations based on this divided-difference calculus are better conditioned numerically and their solutions are numerically stable for in the high-speed processing regime. Thus, we provide a review of this methodology, which is a cornerstone of the general techniques of interest in this study. In the standard formulation of digital signal processing problems, the dynamics of the data and the signal processing elements are described in terms of the classical forward shift operator 5 {ee} = {e+}. (21) 73 From a practical viewpoint, this operator is undesirable as a means of dynamical representation for rapidly sampled signals, because it becomes essentially an identity operator. This means ‘that finite wordlength implementations of systems represented with the shift operator waste valuable bits of accuracy in representing dynamical behavior that is trivial. That is, in the high-speed regime, the data dynamics are better captured in the deviation from the identity, and this is where the rumerical accuracy of finite wordlength processors should be concentrated. ‘This objective can be formalized by replacing the shift operator with an incremental difference operator (or delta operator) defined by 6 = (q~ 1)/A where A is the sampling interval. Thus, oan} = {BE} 2) Note that this operator is a numerical derivative, and as such it is an approximation to the basic dynamical element in continuous time - namely, the forward derivative. Thus, this is a natural framework within which to consider high-speed DSP problems. Of course, since the 6 and q operators ate related by a onc-to-one mapping, signal processing solutions for the 6 formulation are equivalent to those based on the traditional shift operator q from a theoretical viewpoint. However, as noted above, 6-based algorithms are be better conditioned numerically and (unlike shift-operator based algorithms) are numerically stable for high-speed processing envelopes. ‘The incremental difference operator approach to high-speed eampling problems has beon studied for a number of algorithms in digital signal processing, and a survey of this methodology can be found in [27]. (For example, the two particular problems noted above - Kalman-Bucy filtering and autoregressive model fitting and prediction - have been studied in [87] and (116, 114], respectively.) In this section, we will review this methodology, focusing primarily on the specific problem of finite-length linear modeling and prediction. This problem - known as the Levinson problem - represents ubiquitous problem in communications signal processing, and thus it serves well to illustrate the techniques of high-speed processing based on incremental difference operators Traditional fast algorithms for fitting finite-length linear models (J.e., autoregressions) to digital signals include the Levinson-Durbin [72] and Schur [125] algorithms. When these algo- ‘hms are used to solve the Levinson problem, large computational errors can occur due to the problems noted above. In this case, these problems can be traced to the ill-conditioning of the Toeplitz data covariance matrix (which these algorithms are trying to invert) in this high- speed regime. Unfortunately, most fast computational algorithms for digital signal processing are based specifically on the shift-operator formalism, which gives rise to this same Toeplitz covariance structure. Thus, the main challenge in applying the incremental difference operator formalism is to develop algorithms that are competitive with ther shift-operator counterparts in terms of complexity. Fast divided-difference counterparts to each of the algorithms noted above - namely, the Levinson-Durbin algorithm and the Schur algorithm - have been developed, and these will be discussed in the sequel. The Standard Levinson Problem ‘The standard Levinson autoregressive modeling problem is concerned with choosing a pa- rameter vector @y = (09,0, 4n,15---;@nn] with ap,o = 1 to minimize the mean-squared prediction error H{e3(¢)} where én(t) is the error in the model Yat DoaneYe-n = elt), te 2, (2.3) = where {¥;}{2._g, iS an observed wide-sense-stationary signal. This formulation is the basis for a large number of DSP applications 7.4 i As is well known, the coefficients minimizing this mean-squared error are the solutions 10 the so-called Yule-Walker equations: 0 Rade = , (4) 0 where Ry is the (n +1) x (a +1) Toeplitz matrix whose i,j element is cys, the |t — J]- lag correlation coefficient of the signal. The Levinson-Durbin algorithm is an algorithm for recursively solving the Yule-Walker equations of successively higher order. This algorithm is given by (72): Q, , 2=0,1,2,. (25a) with initialization ay = 1, where, for each positive integer k, Ij, denotes the k x k identity matrix and J, denotes the k x k matrix that has all zero entries except for 1’s in its anti-diagonal. The reflection coefficients {7m} are given by net = On / Tn (2.55) where em = [0)-.-50,1] Ragt lus} , (2.50) and tm = Ee} = {1,0,..,0 Rn ga (2.5) ‘The mean-squared error sequence {an} satisfies the recursion M41 = Th ~— 2 [me , 2 = 01,0. (2.5e) with initialization mo co. ‘The numerical stability of the Levinson-Durbin algorithm for solving (2.4) has been estab- lished by Cybenke in [16]. However, as pointed out in [16], in many cases of practical interest the matrix Rp, is ill-conditioned, which results in unacceptable errors when the algorithm is imple- mented. This ill-conditioning occurs when the prediction error x, is very small, or, equivalently, when the reflection coefficients are close to +1. An important case where ill-conditioning of this nature occurs is when the discrete-time signal of interest is obtained by sampling a continuous- time process at feirly rapid rates, since lima j= co for all & and J, if the underlying continuous-time process {Y(t);t € 2} is sufficiently smooth (mean-square continuous). (Here, and in the sequel, A denotes the sampling interval used to produce the discrete-time signal under study.) Since the problem is due to the poor conditioning of Rj, it carinot be solved by using alternative algorithms, like the Schur algorithm, to solve (2.4), as noted in [16] and by Yagle and Levy in [125] ‘Moreover, in this sampled-data case, the following result can be proved. Proposition 2.1 (Vijayan, et al, [116]) Assume that the continuous-time process sa following conditions: i.) {¥ (tit © R} has n—1 mean squace derivatives. 75 ii.) The random vector of derivatives [¥("-9(Q), ...,¥(0), ¥(0)] has a non-singular co- variance matrix. ‘Then, lim ony = (-1) (3). F=O Lem (2.6) and Jim % = (1) (2.7) ‘Thus we see that, as A —+ 0, if the autocovariance function of the continuous-time process has sufficiently many derivatives, the coefficients obtained by the Levinson algorithm will converge to the binomial coefficients (—1) (* ) independently of the underlying process. This points to a major difficulty with the standard Levinson formulation for finite-length linear modeling; namely, the parameters of this model contain no information about the statistics of the underlying process except in terms that are of higher-order in A. The High-speed Levinson Problem In order to correct the difficulties noted above, we consider the reformulation of the autore- gressive modeling problem (2.3) in terms of the divided-difference, or delta, operator. To do 0, we assume for the remainder of Section 2 that the signal (¥e}Z2—oo 1s obtained by uniformly sampling a continuous time process {¥(t); t € 7) at interval A. ‘A mainstay of discrete-time signal and system modeling is the representation of dynamics in ‘terms of the shift operator q, defined in (2.1). The n**-order autoregressive model (2.3) can be rewritten in terms of the shift operator as Anla)Yerin = en(t), te Z, @8) with An(q) = Dfc1Gn,49"-*. (Here, of course, gé denotes £ repeated applications of g; 1% = Vite) For high-speed processing, it has been demonstrated (see (116, 114, 115]) that the numerical problems plaguing such representations can often be ameliorated by considering alternative formulations based on the use of the delta operator as the fundamental dynamical element as noted above. Recall that the delta operator, 6, for sampling interval A is given by as in (2.2), In this context, it is of interest to consider a model of the form Ds wont) VYitn=m(t), be2 (2.10) where 8, = ( Bn0,Bn,ty---+2nnJ? with Bp,o = 1, and {Up (t) #2-co is the sequence of modeling errors in this n‘*-order model. Note that the iteration of the delta operator is (2.11) so that (2.22) and so forth. (One motivation for considering the model (2.10) is its parallelism with the continuous time autoregressive mode! given by dVP"YD(t) + Giga ¥ (tdi +... +dan¥(Qdt = dwt) where (W(t); t € R] is a Brownian motion. Another continuous-time model that has been used in [19, 41, 42] is the following, which is based on an integral operato! avy [ piPiT ~(¢~s))a¥ (8) = dV (2.13) In [19], this model is approximated by using the standard discrete-time AR model with order n=T/A. As A -+ 0, n+ co and the limiting values of the discrete AR parameters any are related to the continuous AR function a(T;t). The disadvantage of this approach is that, for small A, the number of parameters in the model becomes very large (the number n of parameters should grow at a A~! rate). In comparison, (2.10) gives a parsimonious parametrization that also converges to a continuous-time model. Note that the variables &"¥;,6""1Y,,...,Y are obtained by linear transformation of the variables g"Yi.,q"~1¥,,..., Ye. Since &* = (q — 1)*/A*, this transformation can be represented by Tr, an (n+ 1) x (n-+1) matrix whose |, !* element is given by aayh* oak (Trdie = Qe L- as On.) Thus, Ty, is an invertible lower triangular matrix whose n x n right lower submatrix is Ty-1, with Ty = 1. The inverse of this matrix is given by (Ea )e = atte), o

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